Historic advances in computer technology have made it economical for individual users to have their own computing system, which caused the proliferation of the Personal Computer (PC). Continued advances of this computer technology have made these personal computers very powerful but also complex and difficult to manage. For this and other reasons there is a desire in many workplace environments to separate the user interface devices, including the display, keyboard, mouse, audio and the peripheral devices from the application processing parts of the computing system. In this preferred configuration, the user interface devices are physically located at the desktop, while the processing and storage components of the computer are placed in a central location. The user interface devices are then connected to the processor and storage components with some method of communication.
One of the more challenging aspects of separating the user interface from the processing system is the provision of an audio interface at the remote user site with the features and capabilities expected of a desktop PC. Existing remote user interface systems with limited audio capabilities are described below.
The first method is an audio data stream transfer method used by remote desktop applications such as VNC or RDP. Audio information is obtained by the host operating system in conjunction with application software such as Speak Freely and transferred over the communication system using remote desktop protocols to an operating system or application at the remote user interface system. This method is inherently dependent on the host operating system and consequently loads the host data processor with encoding and decoding operations. The operating system is also required to respond to the continuous stream in real time to avoid delays, drop-outs or latency may be detected by the human ear or introduce lip synchronization problems in video applications.
The second method is an audio capture and transfer method. The audio stream is captured only after it has been converted to an analog stream. An application on the host data processor digitizes the audio stream, compresses it and transfers it over the network to an application running on a remote computer, where it is played out through the remote computer's audio drivers and hardware. This introduces a number of delays, operating system dependencies and performance impact on other applications executing on the data processor. In order for full duplex capability to be supported, the method also requires equivalent digitization circuitry and software application support at the remote user system to transfer audio streams back to the host. Another problem with the audio capture and transfer method is that the audio codec and supporting analog circuitry must have extended capabilities to allow any supported codec to operate at optimal performance in terms of channel number, frequency response, dynamic range etc. For example, an audio signal from a microphone input has significantly different characteristics to a CD quality audio line driver. PC audio circuit normally uses an analog audio mixer between the digital to analog converter (DAC) and the analog to digital converters (ADC) to support the wide dynamic range. The circuitry used by the audio capture and transfer method does not have the analog mixing capability and therefore is not capable of supporting the wide dynamic range needed for high quality audio.
Both of the methods described above are reliant on application software or operating system functions at the remote user interface to enable audio functions. However, much of the motivation behind separating the remote user from the data processor is to reduce the complexity and software dependence of the remote user in order to reduce maintenance overheads and equipment costs. These methods do not achieve this objective.
The third method is an audio physical layer extension method that can be found with KVM audio extensions or audio-over-power line methods that do not rely on software at the remote user. Rather than being digitized, the analog audio signal is amplified or modulated and transmitted over a proprietary analog connection. This method is incompatible with standard networks, requiring additional cabling and increasing equipment and maintenance costs. Furthermore, analog transmission introduces noise to the audio signal and the quality degrades as the cabling is extended, limiting the useful distance over which audio may be transferred.
The fourth method that does not have the distance limitation and signal degradation problems described is the Ethernet audio networking method found in professional grade digital audio mixing environments based on devices like the Cirrus Logic Cobranet™ audio interface. A driver on the data processing system captures digital audio information and communicates it across a dedicated Ethernet network to a client or pier system. This method is supported over standard cabling and in the case of simple client decoders, software at the client system is not required. Systems using this method are limited to layer 2 networking protocols, requiring a pre-defined network topology with dedicated wiring. Although Ethernet is used, proprietary network protocols are used at a higher layer to preserve the real time performance rendering the solution incompatible with corporate LAN infrastructure. Other systems such as Livewire by Axia Audio do use IP over Ethernet but require specialized operating system functions and drivers to meet real-time demands.
In summary, existing methods that support remote user interfaces either require software at the remote system or have limited and noise-susceptible audio connectivity. Other solutions support high quality audio over Ethernet but require customized operating system, application software and hardware architecture at the data processor to support the real-time processing requirements for an audio system. Therefore, a better method for providing an audio connection between the data processor and the remote user interface is needed.